Hi Greyman, Should it look like this now? Can i use 2 SIP Proxies just to make sure i have a backup? will that cause any problem again with regards to calling extension to extension? Extensions will register on the asterisk still? How about outbound calls to other SIP provider or a telcobridge, SIP proxy will handle that also? Basically asterisk will ask SIP proxy of everything? Will RTP stream still go thru asterisk?
Also, i plan on setting these up as a Virtual PBX for multiple offices, which means company A can only use Trunks for A, B is for Trunk B etc etc. Does outbound to trunks have any issues? or problem is just basically calling extension to extension? [other voip provider] [telcobridge] -- [pstn] | | -------------------------------------------------------------------- [ SIP Proxy ] -------------------------------------------------------------------- | | | | [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | -------------------------------------------------------------------- | mysql cluster | -------------------------------------------------------------------- Thanks Regards, Ron Grey Man wrote: > On Fri, Feb 29, 2008 at 2:01 AM, Ron <[EMAIL PROTECTED]> wrote: >> Hi All, >> >> If i have this kind of setup, what do i need to make it's load balance. >> >> [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] >> | | | | >> --------------------------------------------------------------------- >> | mysql cluster | >> --------------------------------------------------------------------- >> >> I plan on doing it via DNS SRV only, but if a user register on asterisk >> 1 how can users at asterisk 4 reach that user. Thank You >> >> Regards, >> Ron >> > > Hi Ron, > > If you're using realtime each Asterisk server will know where every > user is irrespective of which Asterisk server they registered on. The > problem is NAT, if a client is behind NAT and registers on server 1 > then server's 2,3 & 4 are unlikely to be able to get through to it. > Last time I lookedthe Asterisk realtime engine doesn't record which > server an account registered on in the database so the only option I > can think of would be to forward an incoming call for a user to all 4 > of your Asterisk servers that way the call will get through but if > they are not behind NAT they'll get 4 incoming calls. > > Bascially it's messy using the set up you've got. What you really need > is a SIP Proxy (assuming you're using SIP, if not it's even trickier). > The SIP Proxy could load balance requests across your Asterisk > servers. For calls destined for your users you can use the > outboundproxy field in the sippeers table, by setting it to the IP > address of your SIP Proxy server you can get Asterisk to forward all > requests for a SIP account through the proxy (there is also an > outboundproxyport setting but avoid it as it's been broken forever). > > There are a few gotchas with a SIP Proxy the main one being transfers. > But if you can get away with not allowing transfers then you are best > to do so as the CDR's Asterisk produces are wrong anyway. > > Regards, > > Greyman. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users